1. Field of the Invention
The present invention pertains to achieving optimal quality when transmitting voice data over a lossy network; more particularly, it pertains to managing jitter buffering of data packets over a packet-switched network.
2. Related Art
Latency and jitter are important aspects of network performance that can degrade communication between any two points on a packet-switched network, like the Internet. Latency is the delay introduced on packets during travel from one site to another. Latency will be perceived by the end users as a delay in the response of the remote site. Jitter is the variation in latency from one packet to another.
Latency and jitter each impact communication differently. For example, if packets always arrived 50 milliseconds (ms) after being transmitted, then there would be a 50 ms latency and no jitter. In another example, however, if packet # 1 arrived 100 ms after transmission, packet # 2 arrived 50 ms after transmission, and packet # 3 arrived 150 ms after transmission, there would be an average jitter of +/−33 ms. In voice over Internet protocol (VOIP) applications, jitter is more critical than latency. Jitter can cause a packet to arrive too late to be useful. The effect is that the packet may be delayed enough that the end user will hear a pause in the voice that is talking to them, which is very unnatural if it occurs during the middle of a word or sentence.
Jitter typically occurs when the network utilization is too high, and packets are being queued, causing delivery times to become unpredictable. The Internet, because of its complex structure, is often subject to varying degrees of jitter. Jitter variation can occur at different locations and at different times depending upon network traffic and other conditions. Thus, jitter needs to be managed.
Effective jitter management is especially needed in VoIP applications. Each VoIP call needs jitter management. FIG. 1 shows an example VoIP architecture 100, including gateways 110, 120 that provide an interface between public-switched telephone networks (PSTN) 130, 140 and a packet-switched network 102. A voice call is carried out between telephone 150 and telephone 160 through PSTN 130, gateway 110, network 102, gateway 120, and PSTN 140.
Static jitter buffering is one conventional technique to compensate for jitter. As shown in FIG. 2, static jitter buffering is carried out in gateway 120 which receives voice packets from network 102. A static jitter buffer 220 is provided to buffer the received voice packets from network 102. In such static jitter buffering, however, there is a compromise between the size of the jitter buffer and the delay of voice packets waiting in the jitter buffer. In particular, if the jitter buffer is large, it accommodates greater variation in jitter. The output packet traffic may not be jittery, but noticeable delays occur. If the jitter buffer is small, the delay is smaller but gaps in traffic are not accommodated.